I'm using Asterisk 1.2.0 source which I downloaded from the Digium website yesterday. I was previously using Asterisk 1.0, and this voice changer was my excuse to upgrade.

I had to delete the /var/lib/asterisk folder because I got some errors during the install process. For some reason, there was a problem overwriting my sound files. I did not delete /etc/asterisk because I figured that wouldn't be touched.

The feature you requested has already been implemented, it will be released in 0.3 within a week or so.

Do my contributors and I still get the beer bounty?

Jman, I believe the hunk that is failing is the part where it adds app_voicechanger.so to the APPS list in apps/Makefile. You can do that manually and it should hopefully work... The patch is REALLY simple.

The feature you requested has already been implemented, it will be released in 0.3 within a week or so.

Do my contributors and I still get the beer bounty?

Jman, I believe the hunk that is failing is the part where it adds app_voicechanger.so to the APPS list in apps/Makefile. You can do that manually and it should hopefully work... The patch is REALLY simple.

It will toggle the modulation in real time inside of a call? Shit yeah, you'll get beer money!

How long have you been working on this? Have you done anything with the people looking to implement a "lie detector" based on vocal stress levels?

What would be REALLY cool is if you could invert the audio frequencies, so only someone with a similar setup could hear the sound correctly.

It will toggle the modulation in real time inside of a call? Shit yeah, you'll get beer money!

How long have you been working on this? Have you done anything with the people looking to implement a "lie detector" based on vocal stress levels?

What would be REALLY cool is if you could invert the audio frequencies, so only someone with a similar setup could hear the sound correctly.

In the next day or so, I'm going to setup a beer fund on the website; after all, it's a lot cooler than a just asking for money. I'll probably split the money 50% me, and 25% for each of my two contributors. As a gentleman, you have my word that every dollar you donate will go towards boozehoundery.

I've been working around my demanding day job schedule the past couple weeks to make the voice changer. The motivation to create the voice changer originated from a prank I had been pulling on a friend. It's a long story I'll jot down on the site one of these days.

As for inverting frequencies, do you mean having a reverse effect to change someone's voice to normal? Claude and I have been thinking about creating a res module for asterisk that would allow you to enter a command to the CLI to manipulate voice on an existing channel. If you got a phone call from someone using the voice changer, that would allow you to go in CLI and revert their voice, assuming that they're not using the random distortion feature I plan to implement. Right now it's more on the low priority end due to a couple factors:

- It will be impossible to install a voice changer on a voip bridged channel- It will be difficult; and I'm not sure how to do it without hacking the * core

I tried downloading asterisk again, and this time the patch worked right off the bat. I'm not sure what was causing the problem earlier, but it works now. I've had some fun scaring my family members already.

I tried downloading asterisk again, and this time the patch worked right off the bat. I'm not sure what was causing the problem earlier, but it works now. I've had some fun scaring my family members already.

Thanks!

Damn, I don't know if it's the syntax that's changed in the 1.20 upgrade of Asterisk, or else if my declaration of a cid is globally overriding my asterisk setup, but now my cid _always_ says Unknown caller.

Notice that the parthentsis after Set is being removed. Also, be aware that if you have the fromuser field set in your sip.conf file, that will override any caller id that you try to send. Make sure that you can normally set caller id, and its not the voice changer screwing things up.

However, when I call my IVR Auto Attendant from my mobile phone to forward through the voicechangedial, it places the call like normal but passes the CPN from my mobile phone as the cid for the call...

I don't have any time to look right now, but I think that you have to set your caller id to null, and then set it as what you'd like to send. I read something about this on voip-info.org, so try checking there.

As for inverting frequencies, do you mean having a reverse effect to change someone's voice to normal?

I believe that they're talking about frequency inversion voice scrambling. Essentially you mix the voice (multiply, not add) with a pilot frequency. This has the effect of inverting the component frequencies of an audio sample. If you're familiar with FFTs then imagine doing an FFT, reversing all of the elements of the FFT, and then performing an IFFT.

Several cheap cordless phones do this to keep out the casual scanner listener. With a tiny bit of code it can be defeated.

I believe that they're talking about frequency inversion voice scrambling. Essentially you mix the voice (multiply, not add) with a pilot frequency. This has the effect of inverting the component frequencies of an audio sample. If you're familiar with FFTs then imagine doing an FFT, reversing all of the elements of the FFT, and then performing an IFFT.

Several cheap cordless phones do this to keep out the casual scanner listener. With a tiny bit of code it can be defeated.

Despite the fact that the patch says 0.2, it is actually 0.2.1. I was too lazy to update the filename.

Wow, voice scrambling is cool! As soon as we get CDR logging in VoiceChangeDial, I will write ScrambleDial. Thanks so much for that link. Any other cool phreaking stuff you guys have up your sleaves that I could write in to Asterisk?